1. Field of the Invention
The present invention relates to an A/D converter apparatus and a D/A converter apparatus comprising a digital filter.
2. Description of the Related Art
An A/D and D/A conversion technique is necessary for connecting information society which demands high-speed and high reliability in digitizing information with the natural world in which all the information is expressed by analog signal, and is widely used in various fields such as audio processing, video processing, and analog signal control.
The A/D conversion means a process for sampling analog signals in a discrete time and quantizing amplitude information to convert it into digital signals. However, since the digital signal obtained by performing A/D conversion often includes quantization noise generated from the quantization process and aliasing noise generated from sampling process, a process for removing these noises is typically required. As a means for accomplishing this, a digital filter is generally used.
The D/A conversion means a process for holding the digital signal represented in the discrete time to represent it in a consecutive time domain and smoothly transitioning the amplitude information quantized by a smoothing filter. Since the smoothing filter is realized by an analog circuit, it is in the trade-off relationship with the circuit scale, but is the filter having a relatively smooth frequency transition. Accordingly, in a digital circuit, the neighboring aliasing noise which cannot be removed by the smoothing filter must be removed. Thus, as a means for accomplishing this, a digital filter is used.
The digital filter used in any one of the A/D conversion and the D/A conversion is to remove the noise and extract a needed spectrum. According to the form of the digital filter, there are four kinds of the digital filters such as a low pass filter, a high pass filter, a band pass filter and a band rejection filter. The frequency characteristics of these digital filters can be classified into three domains such-as a passband, a transition band and a stopband and the form of each domain is determined by the structure and the order of the filter.
As the circuit using the A/D conversion and the D/A conversion, there is an audio CODEC (Coder and Decoder) circuit. The audio CODEC circuit converts an audio analog signal input from a microphone into a digital signal and converts a voice represented with the digital signal into an analog signal, thereby a speaker outputs the voice. In the A/D and D/A conversion technique related to voice, converting a signal component of an audible band without the change thereof and securely removing a signal component out of the audible band are required.
The handling of a frequency component is realized by the digital filter. The more filter is close to an ideal filter (the gradient of the frequency characteristics in the transition band is infinity), the more increased the order of the structure is and the scale of the circuit becomes increased. Accordingly, it is common that the filter is divided into a plurality of stages so that the out-of-band noise component is slowly removed to extract the needed band of frequency component.
FIG. 5 is a schematic diagram of a conventional A/D converter apparatus 500 used in the audio CODEC circuit. The A/D converter apparatus 500 comprises an anti-aliasing filter 1 for previously removing an aliasing noise generated by sampling, an A/D converter 2 for sampling an analog signal at a data rate DR=128Fs (Fs is the sampling frequency of the digital signal which is desired to be obtained), quantizing it and converting it into a digital signal, a SINC filter 3 for removing the aliasing noise due to down-sampling, a decimator 4a for performing the down-sampling at a data rate DR=4Fs, a high pass filter 6 for removing the DC component from the output of the decimator 4a, a low pass filter 5 for removing the aliasing noise generated by down-sampling, and a decimator 4c for performing down-sampling at a data rate DR=Fs.
The sampling rate of the A/D converter 2 should be, for example, at least Fs with respect to the input analog signal of which the spectrum band is limited to ±Fs/2. However, practically, the sampling rate is higher by several times to several tens times, the sampling rate of the digital signal converted by the A/D converter 2 should be reduced by the down-sampling. However, if only down-sampling is simply performed, the aliasing noise and the quantizing noise may be mixed to the needed signal band to remarkably deteriorate the characteristics thereof. Accordingly, the down-sampling must be performed, while removing the signal out-of-band noise.
In FIG. 5, the analog signal (the band thereof is limited to ±Fs/2) input to the A/D converter 2 is sampled at. 128 times (data rate DR=128Fs), and, after the noise in the band of 4Fs is removed by the SINC filter 3 (referred to as a comb filter), the data rate is decreased to 4Fs by the decimator 4a. 
Next, after the DC component of the input audio signal is removed by the high pass filter 6, the audio signal passes through the low pass filter 5 to remove the noise at Fs. Then, the down-sampling is performed by the decimator 4c until the data rate becomes Fs to output the digital data at the data rate of Fs.
In FIG. 5, the decimator 4a performs the down-sampling of the output of the SINC filter so that the data rate is decreased from 128Fs to 4Fs, but the ratio thereof is optional. For example, the down-sampling can be performed till Fs, and, in this case, the sharp SINC filter corresponding thereto may affect the signal band such as attenuating a portion of the needed signal. In order to the compensate it, the frequency characteristics of the low pass filter 5 should finely adjusted. In this case, in order to realize the fine frequency characteristics of the low pass filter 5, a plurality of a design parameters are required and, accordingly, the circuit scale may be increased.
On the contrary, the case that the output of the SINC filter 3 is not decreased to 4Fs can be considered. In this case, since the structure of the SINC filter 3 is simply, the circuit scale of the SINC filter 3 itself is reduced. However, the low pass filter 5 which is the next stage must be sharply realized, and the circuit scale of the low pass filter 5 becomes increased.
Accordingly, the structure of the digital filter and the sampling rate of the data input to the filter are mainly determined by the trade-off with the circuit scale.
However, in the conventional A/D converter apparatus, in case that the characteristics required for the digital filter are changed, if the gradient of two times must be realized in the transition band with respect to the frequency characteristics of the high pass filter 6, the means for satisfying the required characteristics can not be selected by increasing the order of the filter, because the structure of the filter and the data rate are secure.
In the conventional A/D converter apparatus for the audio CODEC circuit, in case that the sharp characteristics is required in the transition band of the high pass filter for removing the DC component, the requirement can be satisfied by increasing the order of the filter, but the circuit scale is increased accompanying with the increment of the order.